1. Field of the Invention
The invention relates to audio signal processes, and in particular, to a codec capable of decoding signals from both analog and digital microphones.
2. Description of the Related Art
Recording capability is prevalent in portable consumer products, such as a mobile phone, a portable media player and a Personal Digital Assistant (PDA). FIG. 1 shows a conventional audio codec 100, an essential component of a digital device (not shown) for processing recorded audio signals. A digital microphone 110 is usually embedded inside the digital device to convert input voices into a second digital signal #D2. The digital microphone 110 may be an Electret Condenser Microphone (ECM) that outputs the second digital signal #D2 in a Pulse Density Modulation (PDM) format. Conventionally, the second digital signal #D2 is a one-bit data stream, and its data rate may vary from an application dependent range, such as 1.024 MHz (for an 8 KHz narrow band communication application) or 3.072 MHz. (for a 48 KHz audio recording).
The digital device may also support externally attached analog microphones, such as a hand free microphone or an ear-microphone. Thus, it is desirable to implement an audio codec 100 that supports analog inputs in addition to the second digital signal #D2. As shown in FIG. 1, the analog input signal #AIN represent an analog input converted from input voices. When an analog microphone (not shown) is attached to the digital device, the analog input signal #AIN is sent from the analog microphone to the audio codec 100. An analog to digital converter (ADC) 102 then converts the analog input signal #AIN into a first digital signal #D1 of the same format as the second digital signal #D2. The first digital signal #D1 and second digital signal #D2 are selected by a multiplexer 104 to be an output for a subsequent stage. For example, when the analog microphone is not available, the multiplexer 104 selects only the second digital signal #D2 to be a selected signal #DS, thus a decimation filter 106 can down sample the selected signal #DS to output a down sampled signal #D′S. Conversely, if the analog microphone is available, the multiplexer 104 may select the first digital signal #D1 to be the selected signal #DS. Next, a data formatter 108 would then convert the selected signal #DS into a predetermined format and output it as a digital output signal #DOUT.
The analog microphone that provides the analog input signal #AIN, however, may feature various characteristics such as signal amplitude, interference and quality. The ADC 102 must be specifically implemented to be adapted to different analog microphones. Conventionally, the ADC 102 may be a sigma-delta modulator, and the first digital signal #D1 produced therefrom is a one-bit data stream with varying pulse densities. Technically, a one-bit data stream requires less wire to transfer, thus the cost can be reduced. The disadvantage of the one-bit form, however, is generally known as being an idle tone of significant magnitude. It is therefore desirable to implement a more flexible architecture that improves signal quality and reduces the idle tone effect.